Question : Unable to call through asterisk

I want to create a voip service.I have installed asterisk-1.4 on a dedicated remotely hosted debian lenny distro. I  made a sip.conf and extensions.conf so as to place a call between two sip phones(i am using xlite 3.0) installed in some other windows Pc. whenever i switch this phones the asterisk console shows that " Registration from '"1000"xx>' failed for '122.168.10.254' - Peer is not supposed to register "
Where xx.xx.xx.xx is the servers ip "
i.e my sip phones are unable to register with the asterisk server.
Please help me to place call between two sip phones

 #sip show peers
Name/username              Host            Dyn Nat ACL Port     Status          
2000                       (Unspecified)    D          0        Unmonitored    
1000                       (Unspecified)    D          0        Unmonitored    
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline]


# sip show registry
Host                            Username       Refresh State                Reg.Time

# sip show channels
Peer             User/ANR    Call ID      Seq (Tx/Rx)  Format           Hold     Last Message
0 active SIP channels
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Sip.conf
 
[general]
context=default
bindport=5060
bindaddr=0.0.0.0
 
 
 
[1000]
secret=1000
type=friend
context=phones
host=dynamic
 
 
 
 
 
[2000]
secret=2000
type=friend
context=phones
host=dynamic
 
extensions.conf
 
[general]
autofallthrough = yes
static = yes
writeprotect = no
clearglobalvars = yes
 
[default]
exten => s,1,Varbose(1|Unrouted call handler)
exten => s,n,Answer()
exten => s,n,Wait(1)
exten => s,n,Playback(tt-weasels)
exten => s,n,Hangup()
 
[incoming_calls]
 
[internal]
exten => 500,1,Verbose(1|Echo test application)
exten => 500,n,Echo()
exten => 500,n,Hangup()
 
exten => 1000,1,Verbose(1|Extension 1000)
exten => 1000,n,Dial(SIP/1000,30)
exten => 1000,n,Hangup()
 
exten => 2000,1,Verbose(1|Extension 2000)
exten => 2000,n,Dial(SIP/2000,30)
exten => 2000,n,Hangup()
 
[phones]
include => internal
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Answer : Unable to call through asterisk

I need you to use the default sip.conf and change the below:


[general]
context=default

bindaddr=Server_IP_Address

disallow=all                    ; First disallow all codecs
allow=ulaw
allow=alaw
allow=g729
allow=gsm

[1000]
secret=1000
type=friend
context=phones
host=dynamic
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